[Asterisk-Users] RTP header compression?

[Leo Ann Boon]

cRTP (compressed RTP, and its official acronym has a small 'c') is done 
at the router level, so it's transparent to the user agent. AFAIK, 
there's no open implementation of cRTP. Cisco switches support it. 
Another approach is ROHC (Robust Header Compression), it works on all IP 
packets. This one is more promising, it's also the standard for 3G 
phones. ROHC can be implemented at end-points like the 3G phones, not 
just routers. There's a Linux ROHC implementation for improving 
throughput on slow links like PPP. See http://rohc.sourceforge.net/

Hope this helps.

Kevin K wrote:

> Uh oh.  I think I may be looking at the wrong tool.
>
> My goal is to implement (in an open source software suite) an 
> RTP/UDP/IP header compression algorithm that would save bandwidth used 
> by voice traffic packets.  So a 5ms G.711 packet that would otherwise 
> be 98 bytes, could be reduced to 62 bytes when its RTP/UDP/IP header 
> is compressed.  This would require me to get into the RTP stack used 
> to packetize the actual data (voice) packets.  From what I am reading, 
> * is not the right software for that kind of work because it don't 
> really touch the voice packets themselves but rather enables the two 
> endpoints to talk to negotiate and connect with each other (as well as 
> utilize a number of other features - vmail, callerID, etc).  Is this 
> correct?
>
> Is what I am looking for an open-sourced SIP user agent?  Or is it 
> more than that?
>
> Any help would be great, as I appear to be confused.
>
> Thanks,
> Kevin
>
> ----Original Message Follows----
> From: John Todd <jtodd at loligo.com>
> Reply-To: asterisk-users at lists.digium.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] RTP header compression?
> Date: Wed, 20 Aug 2003 13:38:22 -0700
>
>> I sent this to the asterisk-dev by accident...
>>
>> ----Original Message Follows----
>>
>> Hi all,
>>
>> I have a couple questions about RTP header compression with Asterisk:
>>
>> 1) Has this been implemented before or is it included in the Asterisk 
>> package?
>> 2) If the answer to (1) is no, is there an RTP stack that this can be 
>> logically implemented into?  Where would that be?
>>
>> Thanks,
>> Kevin
>
>
>
> As I mentioned before in a reply to this:
>
> 1) No.
> 2) Take a look at the source code; you should be able to find the rtp 
> sections in it readily enough.
>
> 3) If you are asking about RTP compression between Asterisk servers, 
> look at IAX2 and trunking mode.  Questions 1 and 2 become less 
> meaningful.
>
> JT
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